similar to: Mediatrix 1204 sip experience?

I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and

I bought a couple of Mediatrix 1204's a few of months back. (Perceived advantages were relatively low overall cost and size per port, and it isn't nearly as vibration sensitive as a PC would be.) Rich Adamson's review from Feb 1 is comprehensive, and the only thing I'd like to add is this: One "feature" of these units that absolutely infuriates me is its behavior for

I added those lines to my configuration, and i just see with ethereal that my client dial and the 1204 led turn on and they started to interchange packets, im newbie with asterisk i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up? could u send me all the configuration i need step by step? ----- Original Message ----- From: "Wojciech

I am attempting to use a Mediatrix 1204 to interface to multizone paging from Asterisk. I have 4 different paging interfaces and want to connect each of those 4 to an FXO port on the Mediatrix. The desired result is to be able to issue some SIP dial string from asterisk, seize the FXO port on the Mediatrix and then have a speech path. I am able to place calls over the Mediatrix when it's

Possible Mediatrix 1204 fxo sip gateway workaround Need some feedback from experienced * users relative to this workaround please please please. Problem: The mediatrix 4-port fxo gateway does not provide any mechanism for * to select which "port" an outbound pstn call will use. (See lots of previous posts over the past four days for more detail if needed.) Our reseller has been

Product Review Mediatrix 1204 4-Port SIP FXO Gateway Firmware: v2.4.10.69 - US Version US Retail: ~$750, Street Price: ~$450. The Mediatrix 1204 SIP FXO gateway is equipped with four RJ11 pstn jacks and one RJ45 Ethernet jack on its rear panel. It terminates the four pstn lines in either Loop Start or Ground Start mode, handles incoming CallerID, and generates either Dial Tone (back towards the

I'm having a problem getting outbound trunking to work using asterisk and an external SIP FXO. 7 digit dialing produces the following output: -- Executing Dial("SIP/mitel-fe17", "SIP/5925660@mediatrix-1204") in new stack -- Called 5925660@mediatrix-1204 -- SIP/mediatrix-1204-645e answered SIP/mitel-fe17 -- Attempting native bridge of SIP/mitel-fe17 and

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

I have successfully implemented 1204 in semi production environment. Just want to share that it works very well, through the firewall (NATed). Unfortunately, it can not register with the server (and authenticate) but otherwise everything is fine. The audio quality is very good. Regards, Wojtek -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie

Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Only very rarely does our call volume exceed three simultaneous connections (inside to

I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with

I have the user manual, I'll send it to your email tonight when I'll be in my home. I have an APA III-4FXO too, until today I can't put it to work with asterisk. Kind regards, Miguel Date: Fri, 03 Sep 2004 16:07:59 +1000 From: Jamie Carl <geek@j-code.net> Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? To:

Hello guys, I need your help related to a mediatrix 1204 configuration. I read some of the messages that you posted in the asterisk users mailing list about the mediatrix 1204 and decided to contact you. I know that the community is not related to Mediatrix devices, but so far I have not found any other group that has work as much as you with them. I bought the mediatrix in Mexico and my provider

Hello All, I am trying to figure out which way to go for a quad port fxo solution with a good echo can on it. My options are the sangoma remora, a mediatrix fxo, or something similar. The issue is that I would need a good EC. This would be on about a 9000 foot loop, and the lines don't function well on a spa-3000 or zaptel tdm 4 port card. Anyone have experience that drives them in a

Dear Group, I have been able to configure my Asterisk BOX to receive calls from Mediatrix 1204. I'm having problems sending calls out via my Mediatrix unit. The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends back a Status : 480 Temporarily Unavailable. This is my configuration on Asterisk; exten => _78996.,1,Dial(SIP/${EXTEN:5}@192.168.6.52) exten =>

Hi, I'm thinking of purchasing a TDM400P card and was wondering if anyone has experienced any echo issues with phones off these cards connecting to the PSTN via the X100P cards? I have had my fingers burnt with a Voip phone > X100P. Cheers, Taff. --------------------------------- ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself -------------- next part

Has anyone made spandsp to work with a digium tdm fxo card? I finally got the rxfax and txfax modules to compile, the spandsp lib installed (and in the libpath), and now receive: -- Starting simple switch on 'Zap/1-1' -- Executing RxFAX("Zap/1-1", "/var/fax.tif") in new stack -- Hungup 'Zap/1-1' I've tried to adjust rxgain/txgain in

Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already